This invention relates to a digital speech interpolation system, and particularly to an efficient digital speech interpolation system in which as many digitized speech signals as possible are transmitted through a transmission line having a limited communication capacity while avoiding freeze-out and assuring a practically satisfactory quality of speech. The term "freeze-out" as used throughout the specification and claims is intended to describe the condition in which the inputs to the digital speech interpolation system overflow the output capacity of such system. This invention is useful particularly for a long distance telephone system such as international telephone lines, because it can widely improve the efficiency of utilization of a transmission line in a digital satellite communication system or a digital undersea cable system.
Owing to highly developed digital signal processing techniques, in telephony a digital speech transmission system is practically used in which speech signals are digitized to be transmitted. For the purpose of economization by efficiently utilizing a transmission line having a limited communication capacity, a digital speech interpolation system called DSI system is employed in this digital speech transmission system. Moreover, a predictive coding system which makes coding with a short bit length is used together with the DSI system. In case of transmitting a plurality of digital speech signal, the DSI system transmits, by detecting sound portions of speech signals in each input trunk, and by combining only said detected sound portion to form new digital signals, the new digital signals through a smaller number of output channels than the number of the input trunks. Generally, digital speech signals are divided into unit blocks which are a ground of a predetermined number of serial samples. For each unit block, a speech detector detects whether or not speech exists in the unit block. The unit blocks in which speech exists are transmitted. On the other hand, in the predictive coding system a predictor predicts a present sample value from past group of sample values of input digital signal. The difference between the predicted value and the actual sample value, i.e., prediction error is calculated with a subtracter. A quantizer performs quantization of prediction error. By abovementioned manner information can be transmitted at a low bit rate. Such typical systems include a delta modulation system which performs coding with one bit and a differential PCM (DPCM) system which performs coding with two or more bits. Among DPCM systems there is an adaptive DPCM (ADPCM) system in which the quantization level interval of the quantizer and the prediction coefficient of the predictor are controlled so as to be of an optimum value at any times.
An efficient DSI system in which a DSI system is combined with a DPCM system or an ADPCM system has been proposed. In the efficient DSI system a very high degree of utilization of transmission line is made possible owing to the effective utilization of transmission line which is inherent in the DSI system transmitting only the speech portions, and owing to the band compression in the predictive coding system transmitting the speech portions at a low bit rate. Namely, by defining a DSI gain as the ratio of the number of DSI input trunks to the number of DSI output channels which ratio is determined by the proportion of the detected and transmitted speech portions to the whole of input signal on the trunk, and by defining a predictive coding gain as a reciprocal of the reduction factor of the number of bits after predictive coding to the number of original coding bits of speech signal, the total gain of the efficient DSI system may be expressed as the product of the DSI gain and the predictive coding gain.
Although in theory a DSI gain of about 2.5 should be obtained because the average operating percentage of speech is generally said to be about 40%, in practice the DSI gain is set to about 2 for safety design to avoid frequent occurrence of freeze-out. If the DSI gain is set to near 2.5, the number of active input trunks of DSI input trunks in which speech is existing would tend to instantaneously exceed the number of DSI output channels, whereby some of the active input trunks could not be connected to an output channel. This would lead to frequent occurrence of freeze-out in which speech is not transmitted. On the other hand, in the predictive coding system, predictive coding with a fixed length of 4 bits is adopted to keep the quality of speech expressed by signal-to quantization noise ratio S/N.sub.q at substantially the same degree as normal 8 to 6 bit PCM. Only a predictive coding gain of at most 2 can be obtained. In this case, the predictive coding maintains redundancy, because 4 bit length coding necessary to low S/N.sub.q speech portions is similarly applied to high S/N.sub.q speech portions.
On the other hand, low speed sampling is useful to effectively utilize a transmitting line. Really, digitization with 6 KHz sampling has been adopted. A 8 KHz sampling is normally adopted for digital speech signal in telephony. This is based on the fact that analog speech signals in telephony are standardized within a transmission frequency band ranging from 0.3 to 3.4 KHz. But, in a speech transmitting system adopting FDM (Frequency Division Multiplex) undersea cable transmitting system, transmission with 3 KHz band has been practically used, so the 6 KHz sampling has been adopted accordingly. More specifically, after speech signals transmitted with 8 KHz-8bit PCM are once converted to 8 KHz-13 bit linear PCM, they are passed through a low pass filter of 3 KHz band and sampling speed is converted to 6 KHz. And the speech signals are predictively coded by a 4 bit quantizing and 6 KHz sampling ADPCM encoder and transmitted to digital undersea cable or digital satellite communication system at 24 kb/s. However, in the speech transmission system with 6 KHz sampling the transmitted frequency band is 0.3 to 3.0 KHz. Therefore, the system has the disadvantage that the high frequency components of 3.0 to 3.4 KHz of the speech signals standardized within the band of 0.3 to 3.4 KHz are cut off, thereby degrading the quality of reproduced speech in its high frequency region.